Audio and speech data are increasingly transported over a packet network such as the Internet with the widespread use of Voice over IP (VoIP) and audio streaming services. A main characteristic associated with the packet network is packet loss. Frequent packet losses can severely degrade speech intelligibility and audio quality.
There are various methods to recover speech or audio signals from lost packets, which broadly can be divided into two classes: sender and receiver based algorithms. Receiver-only approaches, such as G711 Appendix I, require no side information in the packet, however offer only limited performance. Sender-based techniques often employ adding redundancy at the encoder such that the extra information can be utilized at the decoder to fully or partly recover the lost packets. For example, forward error correction (FEC) is one such method that is commonly used. Unfortunately, such methods can significantly increase bandwidth consumption which is highly undesirable when network bandwidth has become more and more precious nowadays.
The approaches described in this section are approaches that could be pursued, but not necessarily approaches that have been previously conceived or pursued. Therefore, unless otherwise indicated, it should not be assumed that any of the approaches described in this section qualify as prior art merely by virtue of their inclusion in this section. Similarly, issues identified with respect to one or more approaches should not be assumed to have been recognized in any prior art on the basis of this section, unless otherwise indicated.